doc/seruser/operation.xml
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 <?xml version="1.0" encoding="UTF-8"?>
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 <!DOCTYPE section PUBLIC "-//OASIS//DTD DocBook XML V4.2//EN" 
    "http://www.oasis-open.org/docbook/xml/4.2/docbookx.dtd">
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 <section id="operation" xmlns:xi="http://www.w3.org/2001/XInclude">
     <sectioninfo>
 	<revhistory>
 	    <revision>
 		<revnumber>$Revision$</revnumber>
 		<date>$Date$</date>
 	    </revision>
 	</revhistory>
     </sectioninfo>
 
     <title>Server Operation</title>
     <section id="operationalpractices">
 	<title>Recommended Operational Practices</title>
 
 	<para>
 	    Operation of a SIP server is not always easy task.
 	    Server administrators are challenged by broken or
 	    misconfigured user agents, network and host failures,
 	    hostile attacks and other stress-makers. All such
 	    situations may lead to an operational failure. It is sometimes
 	    very difficult to figure out the root reason of
 	    a failure, particularly in a distributed environment
 	    with many SIP components involved.		
 	    In this section,
 	    we share some of our practices and refer to tools
 	    which have proven to
 	    make life of administrators easier
 	</para>
 
 	<qandaset>
 	    <qandaentry>
 		<question>
 		    <para>
 			Keeping track of messages is good
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			Frequently, operational errors are discovered or reported
 			with a delay.
 			Users frustrated by an error
 			frequently approach administrators
 			and scream "even though my SIP requests were absolutely ok
 			yesterday, they were mistakenly denied by your server".
 			If administrators do not record all SIP traffic at
 			their site, they will be no more able to identify
 			the problem reason.
 			We thus recommend that site
 			operators record all messages passing their site and keep them
 			stored for some period of time.
 			They may use utilities such as 
 			<application>ngrep 
 			</application> or 
 			<application>tcpdump
 			</application>.
 			There is also a utility <application>
 			    scripts/harv_ser.sh</application> in <application>
 			    ser</application> distribution for post-processing
 			of captured messages. It summarizes messages captured
 			by reply status and user-agent header field.
 		    </para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Real-time Traffic Watching
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			Looking at SIP messages in real-time may help to gain
 			understanding of problems. Though there are commercial
 			tools available, using a simple, text-oriented tool
 			such as <application>ngrep</application> makes the job very well thanks to SIP's textual nature.
 		    </para>
 		    <example id="usingngrep">
 			<title>Using <application>ngrep</application>
 			</title>
 			<para>In this example, all messages at port 5060
 			    which include the string "bkraegelin" are captured
 			    and displayed</para>
 			<programlisting>
 [jiri@fox s]$ ngrep bkraegelin@ port 5060
 interface: eth0 (195.37.77.96/255.255.255.240)
 filter: ip and ( port 5060 )
 match: bkraegelin@
 #
 U +0.000000 153.96.14.162:50240 -> 195.37.77.101:5060
  REGISTER sip:iptel.org SIP/2.0.
  Via: SIP/2.0/UDP 153.96.14.162:5060.
  From: sip:bkraegelin@iptel.org.
  To: sip:bkraegelin@iptel.org.
  Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162.
  Date: Thu, 26 Sep 2002 22:03:55 GMT.
  CSeq: 101 REGISTER.
  Expires: 10.
  Content-Length: 0.
  .
 
 #
 U +0.000406 195.37.77.101:5060 -> 153.96.14.162:5060
  SIP/2.0 401 Unauthorized.
  Via: SIP/2.0/UDP 153.96.14.162:5060.
  From: sip:bkraegelin@iptel.org.
  To: sip:bkraegelin@iptel.org.
  Call-ID: 0009b7aa-1249b554-6407d246-72d2450a@153.96.14.162.
  CSeq: 101 REGISTER.
  WWW-Authenticate: Digest realm="iptel.org", nonce="3d9385170000000043acbf6ba9c9741790e0c57adee73812", algorithm=MD5.
  Server: Sip EXpress router(0.8.8 (i386/linux)).
  Content-Length: 0.
  Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=31604 req_src_ip=153.96.14.162 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==1".
 			</programlisting>
 		    </example>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Tracing Errors in Server Chains
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			A request may pass any number of proxy servers on
 			its path to its destination. If an error occurs
 			in the chain, it is difficult for upstream troubleshooters
 			and/or users complaining to administrators to learn 
 			more about error circumstances. 
 			<application>ser
 			</application> does its best and displays extensive
 			diagnostics information in SIP replies. It allows 
 			troubleshooters and/or users who report to troubleshooters
 			to gain additional knowledge about request processing
 			status. 
 			This extended debugging information is part of the warning 
 			header field. See <xref linkend="usingngrep"/> for an illustration
 			    of a reply that includes such a warning header field. The header
 			    field contains the following pieces of information:
 			    <itemizedlist>
 				<listitem>
 				    <para>
 					Server's IP Address -- good to identify
 					from which server in a chain the reply
 					came.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Incoming and outgoing URIs -- good to
 					learn for which URI the reply was
 					generated, as it may be rewritten
 					many times in the path. Particularly
 					useful for debugging of numbering plans.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Number of Via header fields in replied
 					request -- that helps in assessment of
 					request path length. Upstream clients would
 					not know otherwise, how far away in terms
 					of SIP hops their requests were replied.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Server's process id. That is useful for
 					debugging to discover situations when
 					multiple servers listen at the same
 					address.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					IP address of previous SIP hop as seen by
 					the SIP server.
 				    </para>
 				</listitem>
 			    </itemizedlist>
 		    </para>
 		    <para>
 			If server administrator is not comfortable with
 			disclosing all this information, he can turn them
 			off using the <varname>sip_warning</varname> configuration
 			option.
 		    </para>
 		    <para>
 			A nice utility for debugging server chains is
 			<application>sipsak</application>,
 			SIP Swiss Army Knife, traceroute-like tool for SIP
 			developed at iptel.org. It allows you to send
 			OPTIONS request with low, increasing Max-Forwards 
 			header-fields and follow how it propagates in
 			SIP network. See its webpage at
 			<ulink url="http://sipsak.berlios.de/">
 			    http://sipsak.berlios.de/
 			</ulink>.
 		    </para>
 		    <example>
 			<title>Use of SIPSak for Learning SIP Path</title>
 			<programlisting>
 [jiri@bat sipsak]$ ./sipsak -T -s sip:7271@iptel.org
 warning: IP extract from warning activated to be more informational
 0: 127.0.0.1 (0.456 ms) SIP/2.0 483 Too Many Hops
 1: ?? (31.657 ms) SIP/2.0 200 OK
    without Contact header
 			</programlisting>
 			<para>
 			    Note that in this example, the second hop
 			    server does not issue any warning header fields
 			    in replies and it is thus impossible to display 
 			    its IP address in <application>
 				SIPsak</application>'s output.
 			</para>
 		    </example>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Watching Server Health
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			Watching Server's operation status in real-time may
 			also be a great aid for trouble-shooting. 
 			<application>ser</application> has an excellent 
 			facility, a FIFO server, which allows UNIX
 			tools to access server's internals. (It is 
 			similar to how Linux tool access Linux kernel
 			via the proc file system.) The FIFO server
 			accepts commands via a FIFO (named pipe) and
 			returns data asked for. Administrators do not
 			need to learn details of the FIFO communication
 			and can serve themselves using a front-end
 			utility <application>serctl</application>.
 			Of particular interest for 
 			monitoring server's operation are 
 			<application>serctl</application>
 			commands
 			<command>ps</command> and
 			<command>moni</command>.
 			The former displays running 
 			<application>ser</application>
 			processes, whereas the latter shows statistics.
 		    </para>
 		    <example>
 			<title>serctl ps command</title>
 			<para>
 			    This example shows 10 processes running at a host.
 			    The process 0, "attendant" watches child processes
 			    and terminates all of them if a failure occurs in
 			    any of them. Processes 1-4 listen at local
 			    interface and processes 5-8 listen at Ethernet
 			    interface at port number 5060. Process number
 			    9 runs FIFO server, and process number 10
 			    processes all server timeouts.
 			</para>
 			<programlisting>
 [jiri@fox jiri]$ serctl ps
 0	31590	attendant
 1	31592	receiver child=0 sock=0 @ 127.0.0.1::5060
 2	31595	receiver child=1 sock=0 @ 127.0.0.1::5060
 3	31596	receiver child=2 sock=0 @ 127.0.0.1::5060
 4	31597	receiver child=3 sock=0 @ 127.0.0.1::5060
 5	31604	receiver child=0 sock=1 @ 195.37.77.101::5060
 6	31605	receiver child=1 sock=1 @ 195.37.77.101::5060
 7	31606	receiver child=2 sock=1 @ 195.37.77.101::5060
 8	31610	receiver child=3 sock=1 @ 195.37.77.101::5060
 9	31611	fifo server
 10	31627	timer
 			</programlisting>
 		    </example>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Is Server Alive
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			It is essential for solid operation to know
 			continuously that server is alive. We've been
 			using two tools for this purpose. 
 			<application>sipsak</application>
 			does a great job of "pinging" a server, which
 			may be used for alerting on unresponsive servers.
 		    </para>
 		    <para>
 			<application>monit</application> is
 			a server watching utility which alerts when
 			a server dies.
 		    </para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Dealing with DNS
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			SIP standard leverages DNS. Administrators of
 			<application>ser</application> should
 			be aware of impact of DNS on server's operation.
 			Server's attempt to resolve an unresolvable address
 			may block a server process in terms of seconds. To be
 			safer that the server doesn't stop responding
 			due to being blocked by DNS resolving, we recommend
 			the following practices:
 			<itemizedlist>
 			    <listitem>
 				<para>
 				    Start a sufficient number of children processes.
 				    If one is blocked, the other children will
 				    keep serving.
 				</para>
 			    </listitem>
 			    <listitem>
 				<para>
 				    Use DNS caching. For example, in Linux,
 				    there is an <application>
 					nscd</application> daemon available for
 				    this purpose.
 				</para>
 			    </listitem>
 			    <listitem>
 				<para>
 				    Process transactions statefully if memory
 				    allows. That helps to absorb retransmissions
 				    without having to resolve DNS for each of
 				    them.
 				</para>
 			    </listitem>
 			</itemizedlist>
 		    </para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry id="logging">
 		<question>
 		    <para>
 			Logging
 		    </para>
 		</question>
 		<answer>
 			<para>
 			    <application>ser</application> by default logs
 			    to <application>syslog</application> facility.
 			    It is very useful to watch log messages for
 			    abnormal behavior. Log messages, subject to
 			    <application>syslog</application> configuration
 			    may be stored at different files, or even at remote
 			    systems. A typical location of the log file is
 			    <filename>/var/log/messages</filename>.
 			</para>
 			<note>
 			    <para>
 				One can also use other <application>syslogd</application>
 				implementation. <application>metalog</application>
 				(<ulink url="http://metalog.sourceforge.net/">
 				    http://metalog.sourceforge.net/
 				</ulink>)
 				features regular expression matching that enables
 				to filter and group log messages.
 			    </para>
 			</note>
 			<para>
 			    For the purpose of debugging configuration scripts, one may
 			    want to redirect log messages to console not to pollute
 			    syslog files. To do so configure <application>ser</application>
 			    in the following way:
 			    <itemizedlist>
 				<listitem>
 				    <para>
 					Attach ser to console by setting <varname>fork=no</varname>.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Set explicitly at which address 
 					<application>ser</application>
 					should be listening, e.g., <varname>listen=192.168.2.16</varname>.
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Redirect log messages to standard error by setting
 					<varname>log_stderror=yes</varname>
 				    </para>
 				</listitem>
 				<listitem>
 				    <para>
 					Set appropriately high log level. (Be sure that you redirected logging
 					to standard output. Flooding system logs with many detailed messages
 					would make the logs difficult to read and use.) You can set the global
 					logging threshold value with the option <varname>debug=nr</varname>,
 					where the higher <varname>nr</varname> the more detailed output.
 					If you wish to set log level only for some script events, include
 					the desired log level as the first parameter of the
 					<command>log</command> action in your script.
 					The messages will be then printed if <command>log</command>'s
 					level is lower than the global threshold, i.e., the lower the more
 					noisy output you get.
 					<example>
 					    <title>Logging Script</title>
 					    <programlisting>
 <xi:include href="../../examples/logging.cfg" parse="text"/>
 					    </programlisting>
 					    <para>
 						The following SIP message causes then logging output as shown
 						bellow.
 					    </para>
 					    <programlisting>
 REGISTER sip:192.168.2.16 SIP/2.0
 Via: SIP/2.0/UDP 192.168.2.33:5060
 From: sip:113311@192.168.2.16
 To: sip:113311@192.168.2.16
 Call-ID: 00036bb9-0fd305e2-7daec266-212e5ec9@192.168.2.33
 Date: Thu, 27 Feb 2003 15:10:52 GMT
 CSeq: 101 REGISTER
 User-Agent: CSCO/4
 Contact: sip:113311@192.168.2.33:5060
 Content-Length: 0
 Expires: 600                                 
 					    </programlisting>
 					    <programlisting>
 [jiri@cat sip_router]$ ./ser -f examples/logging.cfg 
 Listening on 
 	192.168.2.16 [192.168.2.16]::5060
 	Aliases: cat.iptel.org:5060 cat:5060 
 WARNING: no fork mode 
 0(0) INFO: udp_init: SO_RCVBUF is initially 65535
 0(0) INFO: udp_init: SO_RCVBUF is finally 131070
 0(17379) REGISTER received
 0(17379) request for other domain received					
 					    </programlisting>
 					</example>
 				    </para>
 				</listitem>
 			    </itemizedlist>
 			</para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			Labeling Outbound Requests
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			Without knowing, which pieces of script code a relayed
 			request visited, trouble-shooting would be difficult.
 			Scripts typically apply different processing to
 			different routes such as to IP phones and PSTN
 			gateways. We thus recommend to label outgoing
 			requests with a label describing the type of processing
 			applied to the request.
 		    </para>
 		    <para>
 			Attaching "routing-history" hints to relayed
 			requests is as easy as using the 
 			<command>append_hf</command>
 			action exported by textops module. The following
 			example shows how different labels are attached
 			to requests to which different routing logic
 			was applied.
 			<example>
 			    <title>"Routing-history" labels</title>
 			    <programlisting>
 # is the request for our domain?
 # if so, process it using UsrLoc and label it so.
 if (uri=~[@:\.]domain.foo") {
     if (!lookup("location")) {
         sl_send_reply("404", "Not Found");
         break;
     };
     # user found -- forward to him and label the request
     append_hf("P-hint: USRLOC\r\n");
 } else {
     # it is an outbound request to some other domain --
     # indicate it in the routing-history label
     append_hf("P-hint: OUTBOUND\r\n");
 };
 t_relay();
 			    </programlisting>
 			    <para>
 				This is how such a labeled requests looks
 				like. The last header field includes
 				a label indicating the script processed
 				the request as outbound.
 			    </para>
 			    <programlisting>
 #
 U 2002/09/26 02:03:09.807288 195.37.77.101:5060 -> 203.122.14.122:5060
  SUBSCRIBE sip:rajesh@203.122.14.122 SIP/2.0.
  Max-Forwards: 10.
  Via: SIP/2.0/UDP 195.37.77.101;branch=53.b44e9693.0.
  Via: SIP/2.0/UDP 203.122.14.115:16819.
  From: sip:rajeshacl@iptel.org;tag=5c7cecb3-cfa2-491d-a0eb-72195d4054c4.
  To: sip:rajesh@203.122.14.122.
  Call-ID: bd6c45b7-2777-4e7a-b1ae-11c9ac2c6a58@203.122.14.115.
  CSeq: 2 SUBSCRIBE.
  Contact: sip:203.122.14.115:16819.
  User-Agent: Windows RTC/1.0.
  Proxy-Authorization: Digest username="rajeshacl", realm="iptel.org", algorithm="MD5", uri="sip:rajesh@203.122.14.122", nonce="3d924fe900000000fd6227db9e565b73c465225d94b2a938", response="a855233f61d409a791f077cbe184d3e3".
  Expires: 1800.
  Content-Length: 0.
  P-hint: OUTBOUND.
 			    </programlisting>
 			</example>
 		    </para>
 		</answer>
 	    </qandaentry>
 	</qandaset>
     </section> <!-- operational practises -->
     
     <section>
 	<title>HOWTOs</title>
 	<para>
 	    This section is a "cookbook" for dealing with common tasks, such as
 	    user management or controlling access to PSTN gateways.
 	</para>
 	<section>
 	    <title>User Management</title>
 	    
 	    <para>
 		There are two tasks related to management of SIP users:
 		maintaining user accounts and maintaining user contacts.
 		Both these jobs can be done using the 
 		<application>serctl</application>
 		command-line tool. Also, the complimentary web
 		interface, <application>serweb</application>,
 		can be used for this purpose as well.
 	    </para>
 	    <para>
 		If user authentication is turned on, which is a highly
 		advisable practice, user account must be created before
 		a user can log in. To create a new user account, call the
 		<command>serctl add</command> utility
 		with username, password and email as parameters. It
 		is important that the environment <varname>SIP_DOMAIN</varname>
 		is set to your realm and matches realm values used in
 		your script. The realm value is used for calculation
 		of credentials stored in subscriber database, which are
 		bound permanently to this value.
 		<screen>
 [jiri@cat gen_ha1]$ export SIP_DOMAIN=foo.bar
 [jiri@cat gen_ha1]$ serctl add newuser secret newuser@foo.bar
 MySql Password: 
 new user added
 		</screen>
 	    </para>
 	    <para><application>serctl</application> can
 		also change user's password or remove existing accounts
 		from system permanently.
 		<screen>
 [jiri@cat gen_ha1]$ serctl passwd newuser newpassword
 MySql Password: 
 password change succeeded
 [jiri@cat gen_ha1]$ serctl rm newuser                
 MySql Password: 
 user removed
 		</screen>
 	    </para>
 	    <para>
 		User contacts are typically automatically uploaded by SIP phones
 		to server during registration process and administrators do not
 		need to worry about them. However, users
 		may wish to append permanent contacts to PSTN gateways
 		or to locations in other administrative domains. 
 		To manipulate the contacts in such cases, use
 		<application>serctl ul</application>
 		tool. Note that this is the only correct way
 		to update contacts -- direct changes to back-end
 		MySql database do not affect server's memory. Also note,
 		that if persistence is turned off (usrloc "db_mode"
 		parameter set to "0"), all contacts are gone on server
 		reboot. Make sure that persistence is enabled if you
 		add permanent contacts.
 	    </para>
 	    <para>
 		To add a new permanent contact for a user, call 
 		<application>serctl ul add &lt;username&gt;
 		    &lt;contact&gt;</application>. To delete 
 		all user's contacts, call 
 		<application>serctl ul rm &lt;username&gt;</application>.
 		<application>serctl ul show &lt;username&gt;</application>
 		prints all current user's contacts.
 		<screen>
 [jiri@cat gen_ha1]$ serctl ul add newuser sip:666@gateway.foo.bar
 sip:666@gateway.foo.bar
 200 Added to table
 ('newuser','sip:666@gateway.foo.bar') to 'location'
 [jiri@cat gen_ha1]$ serctl ul show newuser
 &lt;sip:666@gateway.foo.bar&gt;;q=1.00;expires=1073741812
 [jiri@cat gen_ha1]$ serctl ul rm newuser  
 200 user (location, newuser) deleted
 [jiri@cat gen_ha1]$ serctl ul show newuser
 404 Username newuser in table location not found
 		</screen>
 	    </para>
 	</section> <!-- user management -->
 	<section>
 	    <title>User Aliases</title>
 
 	    <para>
 		Frequently, it is desirable for a user to have multiple
 		addresses in a domain. For example, a user with username "john.doe" wants to be
 		reachable at a shorter address "john" or at a numerical address
 		"12335", so that PSTN callers with digits-only key-pad can reach
 		him too.
 	    </para>
 	    <para>
 		With <application>ser</application>, you can maintain
 		a special user-location table and translate existing aliases to canonical
 		usernames using the <command>lookup</command>
 		action from usrloc module. The following script fragment demonstrates
 		use of <command>lookup</command> for this purpose.
 		<example>
 		    <title>Configuration of Use of Aliases</title>
 		    <programlisting>
 if (!uri==myself) { # request not for our domain...
     route(1); # go somewhere else, where outbound requests are processed
     break;
 };
 # the request is for our domain -- process registrations first
 if (method=="REGISTER") { route(3); break; };
 
 # look now, if there is an alias in the "aliases" table; don't care
 # about return value: whether there is some or not, move ahead then
 lookup("aliases");
 
 # there may be aliases which translate to other domain and for which
 # local processing is not appropriate; check again, if after the
 # alias translation, the request is still for us
 if (!uri==myself) { route(1); break; };
 
 # continue with processing for our domain...
 ...
 		    </programlisting>
 		</example>
 	    </para>
 	    <para>
 		The table with aliases is updated using the
 		<application>serctl</application>
 		tool. <application>
 		    serctl alias add &lt;alias&gt; &lt;uri&gt;</application>
 		adds a new alias, 
 		<application>serctl alias show &lt;user&gt;</application>
 		prints an existing alias, and
 		<application>serctl alias rm &lt;user&gt;</application>
 		removes it.
 		<screen>
 [jiri@cat sip_router]$ serctl alias add 1234 sip:john.doe@foo.bar
 sip:john.doe@foo.bar
 200 Added to table
 ('1234','sip:john.doe@foo.bar') to 'aliases'
 [jiri@cat sip_router]$ serctl alias add john sip:john.doe@foo.bar
 sip:john.doe@foo.bar
 200 Added to table
 ('john','sip:john.doe@foo.bar') to 'aliases'
 [jiri@cat sip_router]$ serctl alias show john                    
 &lt;sip:john.doe@foo.bar&gt;;q=1.00;expires=1073741811
 [jiri@cat sip_router]$ serctl alias rm john  
 200 user (aliases, john) deleted				
 		</screen>
 	    </para>
 	    <para>
 		Note that persistence needs to be turned on in usrloc
 		module. All changes to aliases will be otherwise lost
 		on server reboot. To enable persistence, set the
 		db_mode usrloc parameter to a non-zero value.
 		<programlisting>
 # ....load module ...
 loadmodule "modules/usrloc/usrloc.so"
 # ... turn on persistence -- all changes to user tables are immediately
 # flushed to mysql
 modparam("usrloc", "db_mode",   1)
 # the SQL address:
 modparam("usrloc", "db_url","mysql://ser:secret@dbhost/ser")
 		</programlisting>
 	    </para>
 	</section> <!-- user aliases -->
 	<section id="acl">
 	    <title>Access Control (PSTN Gateway)</title>
 	    <para>
 		It is sometimes important to exercise some sort of
 		access control. A typical use case is when 
 		<application>ser</application> is used
 		to guard a PSTN gateway. If a gateway was not well guarded,
 		unauthorized users would be able to use it to terminate calls in PSTN,
 		and cause high charges to its operator.
 	    </para>
 	    <para>
 		There are few issues you need to understand when
 		configuring <application>ser</application>
 		for this purpose. First, if a gateway is built or configured to
 		accept calls from anywhere, callers may easily bypass your
 		access control server and communicate with the gateway
 		directly. You then need to enforce at transport layer
 		that signaling is only accepted if coming via
 		<application>ser</application> and
 		deny SIP packets coming from other hosts and port numbers.
 		Your network must be configured not to allow forged
 		IP addresses. Also, you need to turn on record-routing
 		to assure that all session requests will travel via 
 		<application>ser</application>.			    
 		Otherwise, caller's devices would send subsequent SIP requests 
 		directly to your gateway, which would fail because of transport 
 		filtering.
 	    </para>
 	    <para>
 		Authorization (i.e., the process of determining who may call where)
 		is facilitated in <application>ser</application>
 		using <emphasis>group membership</emphasis> concept. Scripts make 
 		decisions on whether a caller is authorized to make a call to
 		a specific destination based on user's membership in a group.
 		For example a policy may be set up to allow calls to international
 		destinations only to users, who are members of an "int" group.			    
 		Before user's group membership is checked, his identity
 		must be verified first. Without cryptographic verification of user's
 		identity, it would be impossible to assert that a caller really
 		is who he claims to be.
 	    </para>
 	    <para>
 		The following script demonstrates, how to configure <application>ser</application>
 		as an access control server for a PSTN gateway. The script verifies user
 		identity using digest authentication, checks user's privileges,
 		and forces all requests to visit the server.
 		<example>
 		    <title>Script for Gateway Access Control</title>
 		    <programlisting>
 <xi:include href="../../examples/pstn.cfg" parse="text"/>
 		    </programlisting>
 		</example>
 	    </para>
 	    <para>
 		Use the <application>serctl</application> tool to
 		maintain group membership. 
 		<application>serctl acl grant &lt;username&gt; &lt;group&gt;</application>
 		makes a user member of a group, 
 		<application>serctl acl show &lt;username&gt;</application> shows groups
 		of which a user is member, and
 		<application>serctl acl revoke &lt;username&gt; [&lt;group&gt;]</application>
 		revokes user's membership in one or all groups.
 		<screen>
 [jiri@cat sip_router]$ serctl acl grant john int
 MySql Password: 
 +------+-----+---------------------+
 | user | grp | last_modified       |
 +------+-----+---------------------+
 | john | int | 2002-12-08 02:09:20 |
 +------+-----+---------------------+
 		</screen>
 	    </para>
 	</section> <!-- access control -->
 	<section>
 	    <title>Accounting</title>
 	    <para>
 		In some scenarios, like termination of calls in PSTN, SIP administrators
 		may wish to keep track of placed calls. <application>ser</application>
 		can be configured to report on completed transactions. Reports are sent
 		by default to <application>syslog</application> facility.
 		Support for RADIUS and mysql accounting exists as well.
 	    </para>
 	    <para>
 		Note that <application>ser</application> is no way 
 		call-stateful. It reports on completed transactions, i.e., after 
 		a successful call set up is reported, it drops any call-related 
 		state. When a call is terminated, transactional state for BYE request
 		is created and forgotten again after the transaction completes.
 		This is a feature and not a bug -- keeping only transactional
 		state allows for significantly higher scalability. It is then
 		up to the accounting application to correlate call initiation
 		and termination events.
 	    </para>
 	    <para>
 		To enable call accounting, tm and acc modules need to be loaded,
 		requests need to be processed statefully and labeled for
 		accounting. That means, if you want a transaction to be reported,
 		the initial request must have taken the path 
 		"<command>setflag(X)</command>, <command>t_relay</command>"
 		in <application>ser</application> script. X must have the
 		value configured in <varname>acc_flag</varname>
 		configuration option.
 	    </para>
 	    <para>
 		Also note, that by default only transactions that initiate
 		a SIP dialog (typically INVITE) visit a proxy server.
 		Subsequent transactions are exchanged directly between
 		end-devices, do not visit proxy server and cannot be
 		reported. To be able to report on subsequent transactions,
 		you need to force them visit proxy server by turning 
 		record-routing on. 
 	    </para>
 	    <para>
 		<example>
 		    <title>Configuration with Enabled Accounting</title>
 		    <programlisting>
 <xi:include href="../../examples/acc.cfg" parse="text"/>
 		    </programlisting>
 		</example>
 	    </para>
 	</section> <!-- accounting -->
 	<section>
 	    <title>Reliability</title>
 
 	    <para>
 		It is essential to guarantee continuous
 		service operation even under erroneous conditions, 
 		such as host or network failure. The major issue in such
 		situations is transfer of operation to a backup
 		infrastructure and making clients use it.
 	    </para>
 	    <para>
 		The SIP standard's use of DNS SRV records has been
 		explicitly constructed to handle with server failures.
 		There may be multiple servers responsible for a domain
 		and referred to by DNS. If it is impossible to communicate
 		with a primary server, a client can proceed to another one.
 		Backup servers may be located in a different geographic
 		area to minimize risk caused by areal operational
 		disasters: lack of power, flooding, earthquake, etc.
 		<note>
 		    <sidebar>
 			<para>Unless there are redundant DNS
 			    servers, fail-over capability cannot be guaranteed.
 			</para>
 		    </sidebar>
 		</note>
 		Unfortunately, at the moment of writing this documentation
 		(end of December 2002) only very few SIP products
 		actually implement the DNS fail-over mechanism. Unless
 		networks with SIP devices supporting this mechanism are
 		built, alternative mechanisms must be used to force 
 		clients to use backup servers. Such a mechanism is
 		disconnecting primary server and replacing it with
 		a backup server locally.
 		It unfortunately precludes geographic dispersion and
 		requires network multihoming to avoid dependency on
 		single IP access. Another method is to update DNS
 		when failure of the primary server is detected.
 		The primary drawback of this method is its latency:
 		it may take long time until all clients learn to use
 		the new server.
 	    </para>
 	    <para>
 		The easier part of the redundancy story is replication of 
 		<application>ser</application>
 		data. <application>ser</application>
 		relies on replication capabilities of its back-end database.
 		This works with one exception: user location database.
 		User location database is a frequently accessed table,
 		which is thus cached in server's memory to improve
 		performance. Back-end replication does not affect
 		in-memory tables, unless server reboots. To facilitate
 		replication of user location database, 
 		server's SIP replication feature must be enabled
 		in parallel with back-end replication.
 	    </para>
 	    <para>
 		The design idea of replication of user location database
 		is easy: Replicate any successful REGISTER requests to
 		a peer server. To assure that digest credentials can
 		be properly verified, both servers need to use the same
 		digest generation secret and maintain synchronized time.
 		A known limitation of this method is it does not replicate
 		user contacts entered in another way, for example using
 		web interface through FIFO server.
 		The following script example shows configuration of
 		a server that replicates all REGISTERs.
 		<example>
 		    <title>Script for Replication of User Contacts</title>
 		    <programlisting>
 <xi:include href="../../examples/replicate.cfg" parse="text"/>
 		    </programlisting>
 		</example>
 	    </para>
 	</section> <!-- reliability -->
 	<section>
 	    <title>Stateful versus Stateless Forwarding</title>
 	    <para>
 		<application>ser</application> allows both stateless
 		and stateful request processing. This memo explains what are pros and cons of
 		using each method. The rule of thumb is "stateless for scalability,
 		stateful for services". If you are unsure which you need, stateful
 		is a safer choice which supports more usage scenarios.
 	    </para>
 	    <para>
 		Stateless forwarding with the
 		<command>forward(uri:host, uri:port)</command> action
 		guarantees high scalability. It withstands high load and
 		does not run out of memory. A perfect use of stateless forwarding
 		is load distribution.
 	    </para>
 	    <para>
 		Stateful forwarding using the <command>t_relay()</command>
 		action is known to scale worse. It can quickly run out of memory and
 		consumes more CPU time. Nevertheless, there are scenarios which are
 		not implementable without stateful processing. In particular:
 		<itemizedlist>
 		    <listitem>
 			<para>
 			    <emphasis>Accounting</emphasis> requires stateful processing
 			    to be able to collect transaction status and issue a single
 			    report when a transaction completes.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    <emphasis>Forking</emphasis> only works with stateful forwarding.
 			    Stateless forwarding only forwards to the default URI out of the
 			    whole destination set.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    <emphasis>DNS resolution</emphasis>. DNS resolution may be
 			    better served with stateful processing. If a request is forwarded
 			    to a destination whose address takes long time to resolve,
 			    a server process is blocked and unresponsive. Subsequent 
 			    request retransmissions from client will cause other processes
 			    to block too if requests are processed statelessly. As a result,
 			    <application>ser</application> will quickly
 			    run out of available processes. With stateful forwarding,
 			    retransmissions are absorbed and do not cause blocking of
 			    another process.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    <emphasis>Forwarding Services</emphasis>. All sort of services 
 			    with the "forward_on_event" logic, which rely on 
 			    <command>t_on_failure</command> tm
 			    action must be processed statefully.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    <emphasis>
 				Fail-over.
 			    </emphasis>
 			    If you wish to try out another destination, after a primary destination
 			    failed you need to use stateful processing. With stateless processing
 			    you never know with what status a forwarded request completed downstream
 			    because you immediately release all processing information after the 
 			    request is sent out. 
 
 			    <note>
 				<para>
 				    Positive return value of stateless
 				    <command>forward</command> action only indicates that
 				    a request was successfully sent out, and does not gain any knowledge
 				    about whether it was successfully received or replied. Neither does
 				    the return value of
 				    the stateful <command>t_relay</command> action family
 				    gain you this knowledge. However, these actions store transactional
 				    context with which includes original request and allows you to
 				    take an action when a negative reply comes back or a timer strikes.
 				    See <xref linkend="replyprocessingsection"/> for an example script 
 					which launches another
 					branch if the first try fails.
 				</para>
 			    </note>
 
 			</para>
 		    </listitem>
 		</itemizedlist>
 	    </para>
 	</section> <!-- stateful vs. stateless -->
 	<section>
 	    <title>Serving Multiple Domains</title>
 	    <para>
 		<application>ser</application> can be configured to
 		serve multiple domains. To do so, you need to take the following steps:
 		<orderedlist>
 		    <listitem id="createtable">
 			<para>
 			    Create separate subscriber and location database table
 			    for each domain served and name them uniquely.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    Configure your script to distinguish between multiple
 			    served domains. Use regular expressions for domain
 			    matching as described in <xref linkend="redomainmatching"/>.
 			</para>
 		    </listitem>
 		    <listitem>
 			<para>
 			    Update table names in usrloc and auth actions to reflect
 			    names you created in <xref linkend="createtable"/>.
 			</para>
 		    </listitem>
 		    
 		</orderedlist>
 	    </para>
 	    <para>
 		The latest <application>SER</application> release includes automated
 		multidomain management which greatly automates maintenance of multiple	
 		domains. Ask our technical support for more help.
 	    </para>
 	</section> <!-- multiple domains -->
 	<section id="missedcalls">
 	    <title>Reporting Missed Calls</title>
 	    <para>
 		<application>ser</application> can report missed
 		calls via <application>syslog</application> facility
 		or to mysql. Mysql reporting can be utilized by 
 		<application>ser</application>'s 
 		complementary web-interface, <application>serweb</application>.
 		(See more in <xref linkend="serweb"/>).
 	    </para>
 	    <para>
 		Reporting on missed calls is enabled by acc module.
 		There are two cases, on which you want to report. The first
 		case is when a callee is off-line. The other case is when
 		a user is on-line, but call establishment fails. There
 		may be many failure reasons (call cancellation, inactive phone,
 		busy phone, server timer, etc.), all of them leading to
 		a negative (>=300) reply sent to caller. The acc module
 		can be configured to issue a missed-call report whenever
 		a transaction completes with a negative status. Two following
 		script fragment deals with both cases.
 	    </para>
 	    <para>
 		First, it reports
 		on calls missed due to off-line callee status
 		using the <command>acc_request</command>
 		action. The action is wrapped in transactional
 		processing (<command>t_newtran</command>)
 		to guarantee that reports are not
 		duplicated on receipt of retransmissions.
 	    </para>
 	    <para>
 		Secondly, transactions to on-line users are marked
 		to be reported on failure. That is what the 
 		<command>setflag(3)</command> action
 		is responsible for, along with the configuration option
 		"log_missed_flag". This option configures <application>ser</application>
 		to report on all transactions, which were marked
 		with flag 3.			   
 		<programlisting>
 loadmodule("modules/tm/tm.so");
 loadmodule("modules/acc/acc.so");
 ....
 # if a call is labeled using setflag(3) and is missed, it will
 # be reported
 ...
 modparam("acc", "log_missed_flag", 3 );
 if (!lookup("location")) {
     # call invitations to off-line users are reported using the
     # acc_request action; to avoid duplicate reports on request
     # retransmissions, request is processed statefully (t_newtran,
     # t_reply)
     if ((method=="INVITE" || method=="ACK") &amp;&amp; t_newtran() ) {
          t_reply("404", "Not Found");
          acc_request("404 Not Found");
          break;
     };
     # all other requests to off-line users are simply replied
     # statelessly and no reports are issued
     sl_send_reply("404", "Not Found");
     break;
 } else {
     # user on-line; report on failed transactions; mark the
     # transaction for reporting using the same number as 
     # configured above; if the call is really missed, a report
     # will be issued
     setflag(3);
     # forward to user's current destination
     t_relay();
     break;
 };
 		</programlisting>
 		
 	    </para>
 	</section> <!-- missed calls -->
 	<section>
 	    <title>NAT Traversal</title>
 	    <para>
 		NATs are worst things that ever happened to SIP. These devices
 		are very popular because they help to conserve IP address space
 		and save money charged for IP addresses. Unfortunately, they
 		translate addresses in a way which is not compatible with SIP.
 		SIP advertises receiver addresses in its payload. The advertised
 		addresses are invalid out of NATed networks. As a result,
 		SIP communication does not work across NATs without extra
 		effort.
 	    </para>
 	    <para>
 		There are few methods that may be deployed to traverse NATs.
 		How proper their use is depends on the deployment scenario.
 		Unfortunately, all the methods have some limitations and
 		there is no straight-forward solution addressing all
 		scenarios. Note that none of these methods takes explicit
 		support in <application>ser</application>.
 	    </para>
 	    <para>
 		The first issue is whether SIP users are in control of 
 		their NATs. If not (NATs are either operated by ISP or
 		they are sealed to prevent users setting them up), the
 		only method is use of a STUN-enabled phone. STUN is 
 		a very simple protocol used to fool NAT in such a way,
 		they permit SIP sessions. Currently, we are aware of
 		one softphone (kphone) and one hardphone (snom) with
 		STUN support, other vendors are working on STUN support
 		too. Unfortunately, STUN gives no NAT traversal
 		guarantee -- there are types of NATs, so called
 		symmetric NATs, over which STUN fails to work.
 		<note>
 		    <para>
 			There is actually yet another method to address
 			SIP-unaware, user-uncontrolled NATs. It is based
 			on a proxy server, which relays all signaling and
 			media and mangles packets to make them more
 			NAT-friendly. The very serious problem with this
 			method is it does not scale.
 		    </para>
 		</note>
 	    </para>
 	    <para>
 		If users are in control of their own NAT, as typically residential
 		users are, they can still use STUN. However, they may use other
 		alternatives too. One of them is to replace their NAT with
 		a SIP-aware NAT. Such NATs have built-in SIP awareness,
 		that patches problems caused by address translations. Prices
 		of such devices are getting low and there are available
 		implementations (Intertex, Cisco/PIX). No special support
 		in phones is needed.
 	    </para>
 	    <para>
 		Other emerging option is UPnP. UPnP is a protocol that allows
 		phones to negotiate with NAT boxes. You need UPnP support in
 		both, NAT and phones. As UPnP NATs are quite affordable,
 		costs are not an obstacle. Currently, we are aware of one
 		SIP phone (SNOM) with UPnP support.
 	    </para>
 	    <para>
 		Geeks not wishing to upgrade their firewall to a SIP-aware or
 		UPnP-enabled one may try to configure static address translation.
 		That takes phones with configuration ability to use fixed port
 		numbers and advertise outside address in signaling. Cisco phones
 		have this capability, for example. The NAT devices need to
 		be configured to translate outside port ranges to the 
 		ranges configured in phones.		    
 	    </para>
 	</section> <!-- NAT traversal -->
 
 	<section>
 	    <title>Using Only Latest User's Contact for Forwarding
 	    </title>
 	    <para>
 		In some scenarios, it may be beneficial only to use only one
 		registered contact per user. If that is the case, setting
 		registrar module's parameter <varname>append_branches</varname>
 		to 1 will eliminate forking and forward all requests only
 		to a single contact. If there are multiple contacts, a contact
 		with highest priority is chosen. This can be changed to
 		the "freshest" contact by setting module parameter's
 		<varname>desc_time_order</varname> to 1.
 	    </para>
 
 	</section>
 
 	<section>
 	    <title>Authentication Policy: Prevention of Unauthorized Domain 
 		Name Use in From and More</title>
 	    <para>
 		Malicious users can claim a name of domain, to which they do 
 		not administratively belong, in From header field. This
 		behavior cannot be generally prevented. The reason is
 		that requests with such a faked header field do not need
 		to visit servers of the domain in question. However, if they
 		do so, it is desirable to assure that users claiming
 		membership in a domain are actually associated with it.
 		Otherwise the faked requests would be relayed and appear
 		as coming from the domain, which would increase
 		credibility of the faked address and decrease credibility of
 		the proxy server.
 	    </para>
 	    <para>
 		Preventing unauthorized domain name use in relayed requests 
 		is not difficult.
 		One needs to authenticate each request with name of the
 		served domain in From header field. To do so, one can
 		search for such a header field using <command>search</command>
 		action (textops module) and force authentication if the
 		search succeeds.
 		<note>
 		    <para>
 			A straight-forward solution might be to authenticate
 			ALL requests. However, that only works in closed
 			networks in which all users have an account in the
 			server domain. In open networks, it is desirable to permit
 			incoming calls from callers from other domains without
 			any authentication. For example, a company may wish
 			to accept calls from unknown callers who are
 			new prospective customers.
 			
 		    </para>
 		</note>
 		<programlisting>
 # does the user claim our domain "foo.bar" in From?
 if (search("^(f|From):.*foo.bar")) {
     # if so, verify credential
     if (!proxy_authorize("foo.bar", "subscriber")) { 
          # don't proceed if credentials broken; challenge
          proxy_challenge("foo.bar", "0");
          break;
     };
 };
 		</programlisting>
 	    </para>
 	    <para>
 		In general, the authentication policy may be very rich. You may not
 		forget each request deserves its own security and you need to 
 		decide whether it shall be authenticated or not. As mentioned
 		above, in closed networks, you may want to authenticate absolutely 
 		every request. That however prohibits traffic from users from
 		other domains. A pseudo-example of a reasonable policy is attached:
 		it looks whether a request is registration, it claims to originate
 		from our domain in From header field, or is a local request to
 		another domain.
 		<programlisting>
 # (example provided by Michael Graff on [serusers] mailing list
 if (to me):
     if register
      www_authorize or fail if not a valid register
      done
    if claiming to be "From" one of the domains I accept registrations for
    proxy_authorize
    done
    if not to me (I'm relaying for a local phone to an external address)
     proxy_authorize
    done
 		</programlisting>
 	    </para>
 	    <para>
 		You also may want to apply additional restriction to how
 		digest username relates to usernames claimed in From and
 		To header fields. For example, the <command>check_to</command>
 		action enforces the digest id to be equal to username
 		in To header fields. That is good in preventing someone
 		with valid credentials to register as someone else
 		(e.g., sending a REGISTER with valid credentials of
 		"joe" and To belonging to "alice"). Similarly,
 		<command>check_from</command> is used
 		to enforce username in  from to equal to digest id.
 		<note>
 		    <para>
 			There may be a need for a more complex relationship
 			between From/To username and digest id. For example,
 			providers with an established user/password database
 			may wish to keep using it, whereas permitting users
 			to claim some telephone numbers in From. To address
 			such needs generally, there needs to be a 1:N mapping
 			between digest id and all usernames that are acceptable
 			for it. This is being addressed in a newly contributed
 			module "domain", which also addresses more generally
 			issues of domain matching for multidomain scenarios.
 		    </para>
 		</note>
 	    </para>
 	    <para>
 		Other operational aspect affecting the authentication policy
 		is guarding PSTN gateways (see <xref linkend="acl"/>). There
 		    may be destinations that are given away for free whereas
 		    other destinations may require access control using
 		    group membership, to which  authentication is a prerequisite.
 	    </para>
 
 	</section> <!-- authentication policy, faked froms -->
 	<section>
 	    <title>Connecting to PBX Voicemail Using a Cisco Gateway</title>
 	    <para>
 		In some networks, administrators may wish to utilize their
 		PBX voicemail systems behind PSTN gateways. There is a practical problem
 		in many network settings: it is not clear for whom a call to
 		voicemail is. If voicemail is identified by a single number,
 		which is then put in INVITE's URI, there is no easy way to
 		learn for whom a message should be recorded. PBX voicemails
 		utilize that PSTN protocols signal the number of originally
 		called party. If you wish to make the PBX voicemail work,
 		you need to convey the number in SIP and translate it in
 		PSTN gateways to its PSTN counterpart.
 	    </para>
 	    <para>
 		There may be many different ways to achieve this scenario. Here
 		we describe the proprietary mechanism Cisco gateways use and how to 
 		configure <application>ser</application> to
 		make the gateways happy. Cisco gateways expect the number
 		of originally called party to be located in proprietary
 		<varname>CC-Diversion</varname> header field. When a SIP 
 		INVITE sent via a PSTN gateway to PBX voicemail has number
 		of originally called party in the header field, the voicemail
 		system knows for whom the incoming message is. That is at least
 		true for AS5300/2600 with Cisco IOS 12.2.(2)XB connected to
 		Nortel pbxs via PRI. (On the other hand, 12.2.(7b) is known
 		not to work in this scenario.)
 	    </para>
 	    <para>
 		<application>ser</application> needs then to
 		be configured to append the <varname>CC-Diversion</varname>
 		header field name for INVITEs sent to PBX voicemail.
 		The following script shows that: when initial forwarding
 		fails (nobody replies, busy is received, etc.), a new branch
 		is initiated to the pbx's phone number. 
 		<command>append_urihf</command> is used to
 		append the <varname>CC-Diversion</varname> header field. It
 		takes two parameters: prefix, which includes header name,
 		and suffix which takes header field separator. 
 		<command>append_urihf</command> inserts
 		original URI between those two.
 		<example>
 		    <title>Forwarding to PBX/Voicemail via Cisco Gateways</title>
 		    <programlisting>
 <xi:include href="../../examples/ccdiversion.cfg" parse="text"/>
 		    </programlisting>
 		</example>
 		
 	    </para>
 	</section>
     </section> <!-- howtos -->
 
     <section>
 	<title>Troubleshooting</title>
 	<para>
 	    This section gathers practices how to deal with errors
 	    known to occur frequently. To understand how to watch
 	    SIP messages, server logs, and in general how to
 	    troubleshoot, read also <xref linkend="operationalpractices"/>. 
 	</para>
 	<qandaset>
 	    <qandaentry>
 		<question>
 		    <para>
 			SIP requests are replied by <application>ser</application> with
 			"483 Too Many Hops" or "513 Message Too Large"
 		    </para>
 		</question>
 
 		<answer>
 		    <para>
 			In both cases, the reason is probably an error in
 			request routing script which caused an infinite loop.
 			You can easily verify whether this happens by
 			watching SIP traffic on loopback interface. A typical
 			reason for misrouting is a failure to match local
 			domain correctly. If a server fails to recognize
 			a request for itself, it will try to forward it
 			to current URI in believe it would forward them
 			to a foreign domain. Alas, it forwards the request
 			to itself again. This continues to happen until
 			value of max_forwards header field reaches zero
 			or the request grows too big. Solutions is easy:
 			make sure that domain matching is correctly
 			configured. See <xref linkend="domainmatching"/>
 			    for more information how to get it right.
 		    </para>
 		</answer>		    
 	    </qandaentry>
 	    <qandaentry id="msmbug">
 		
 		<question>
 		    
 		    <para>
 			
 			Windows Messenger authentication fails.
 		    </para>
 		</question>
 		<answer>
 			<para>
 			    The most likely reason for this problem is a bug
 			    in Windows Messenger. WM only authenticates if
 			    server name in request URI equals authentication
 			    realm. After a challenge is sent by SIP server,
 			    WM does not resubmit the challenged request at all
 			    and pops up authentication window again.
 			    If you want to authenticate WM, you need to
 			    set up your realm value to equal server name.
 			    If your server has no name, IP address can be used
 			    as realm too. The realm value is configured in
 			    scripts as the first parameter of all
 			    <command>{www|proxy}_{authorize|challenge}</command>
 			    actions.
 			</para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry id="mhomed">
 		<question>
 		    <para>
 			On a multihomed host, forwarded messages carry other 
 			interface in Via than used for sending, or messages 
 			are not sent and an error log is issued "invalid 
 			sendtoparameters one possible reason is the server 
 			is bound to localhost".
 		    </para>
 		</question>
 		<answer>
 			<para>
 			    Set the configuration option <varname>mhomed</varname>
 			    to "1". <application>ser</application>
 			    will then attempt to calculate the correct interface.
 			    It's not done by default as it degrades performance
 			    on single-homed hosts or multi-homed hosts that are
 			    not set-up as routers.
 			</para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			I receive "ERROR: t_newtran: transaction already in process" in my logs.
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			That looks like an erroneous use of tm module in script.
 			tm can handle only one transaction per request. If you
 			attempt to instantiate a transaction multiple times,
 			<application>ser</application> will complain.
 			Anytime any of <command>t_newtran</command>,
 			<command>t_relay</command> or 
 			<command>t_relay_to_udp</command> actions is
 			encountered, tm attempts to instantiate a transaction.
 			Doing so twice fails. Make sure that any of this
 			commands is called only once during script execution.
 		    </para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			I try to add an alias but 
 			<command>serctl</command>
 			complains that table does not exist.
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			You need to run <application>ser</application>
 			and use the command
 			<command>lookup("aliases")</command>
 			in its routing script. That's because the table 
 			of aliases is
 			stored in cache memory for high speed. The cache
 			memory is only set up when the 
 			<application>ser</application>
 			is running and configured to use it. If that is
 			not the case, 
 			<application>serctl</application>
 			is not able to manipulate the aliases table.
 		    </para>
 		</answer>
 	    </qandaentry>
 
 	    <qandaentry>
 		<question>
 		    <para>I started <application>ser</application> with
 			<varname>children=4</varname> but many more processes
 			were started. What is wrong?
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			That's ok. The <varname>children</varname> parameter defines
 			how many children should process each transport protocol in
 			parallel. Typically, the server listens to multiple protocols
 			and starts other supporting processes like timer or FIFO
 			server too. Call <application>serctl ps</application> to watch
 			running processes.
 		    </para>
 		</answer>
 	    </qandaentry>
 	    <qandaentry>
 		<question>
 		    <para>
 			I decided to use a compiled version of <application>ser</application>
 			but it does not start any more.
 		    </para>
 		</question>
 		<answer>
 		    <para>
 			You probably kept the same configuration file, which tries to load modules
 			from the binary distribution you used previously. Make sure that modules
 			paths are valid and point to where you compiled <application>ser</application>.
 			Also, watch logs for error messages "ERROR: load_module: could not open 
 			module".
 		    </para>
 		</answer>
 	    </qandaentry>
 	    
 	</qandaset>
     </section> <!-- troubleshooting -->
 </section>